Scenario

You are setting up your keyboard (or any instrument, really) for use with an external sequencer program (Steinberg Cubase, Apple Logic, MOTU Digital Performer, Ableton Live, etc.), and you are unsure how hot to run your gain levels for recording and for mixdown.

Disclaimer

While the factual statements made in this article aren't likely to be disputed, not every experienced engineer/producer will agree with some of the specific recommendations. The original author of this article has been involved in audio engineering and production at a semi-pro level for 12+ years and has experienced first hand the move from a purely analog realm into the various iterations of the digital realm so far. But hey, I'm not perfect and you might well disagree with some of what I'm about to assert. At the end of this article are some links to essays by other "experts" who have done a lot to educate people in some of the nuances of modern digital recording, mixdown, and pre-mastering.

Closely-related articles

The executive summary

If you want to avoid reading a lot of details, the short story is this:

Record at 24-bits. 16-bits is antiquated now and doesn't give you enough dynamic range and headroom to work with.

The analog input signal from the M3 into your audio interface should be loud enough to start causing red flashes on the peaks (in your audio interface's input meters), and then you should drop the volume just enough to make the red flashes disappear.

The digital signal from your audio interface into your sequencer's individual recording tracks should be set at -12 to -18dB on the input meters for each track being recorded. Don't go near -6dB and certainly don't get anywhere near 0dB when tracking.

The digital signal in your sequencer's Master meter should peak at -6dB when creating your mixdown. If you go over -6dB peak on the Master, lower the volume of all your tracks slightly, and equally, to reduce the summed volume of all tracks as needed to keep the Master peaking at -6dB

Render your mixdown to a WAV or AIFF file still at -6dB because normalization and limiting from within all major sequencers is pretty weak.

If you plan to burn your own CDs or encode your own downloadable files (MP3s, etc.), run your rendered file through SoundForge or Wavelab and use their much better limiters or normalization utilities to raise the volume of your WAV or AIFF file to roughly -0.2dB, with about -3dB of limiting on the peaks in the file. Don't overcompress your material just to compete in the Loudness War because that's a race to the bottom and creates mushy junk out of your beautiful mixdown. You'll be in the ballpark, loudness-wise, if you started with a good mixdown and normalize to -0.2dB with moderate -3dB compression.

Older advice for analog equipment is BAD for digital recording

While some books and magazines and web-based essays about recording have gotten better over the past 5 years, there are still many resources out there that still give advice from the perspective of recording with (mostly) analog gear. The problem is that good rules of thumb for setting gain levels on analog equipment are in fact very bad rules of thumb when you are recording and mixing in the mostly digital realm.

The main problem if you are following advice meant for a (mostly) analog recording, which was common in the early days of digital recording too (the ADAT and Tascam days), revolves around the concepts of "signal to noise ratio" and "headroom":

Signal to noise (SN) ratio is how much space there is between the lowest signal levels in which you hear electronic background noise and the average RMS level (essentially the average volume perceived by your ears) of your recorded signal. The higher the SN ratio, the "cleaner" the recorded signal. If the SN ratio is too low, then when you increase the volume of such a track to make it sit at the right spot in your mix, you also increase the level of the audible background noise in the track.

In the analog realm, getting the highest possible SN ratio without clipping your peaks too much is very desirable, because it results in the least amount of background noise in track that can come back to haunt you later.

In the digital realm, there is no real noise floor. So SN ratio doesn't really come into play once you're working with digital signals. Instead, the concept of "resolution" (which relates to the bit-depth of your digital waveform) comes into play and can add a kind of mathematical artifact-created noise if your signal is extremely low, but in most practical situations you won't run into this, especially when working at the current defacto standard of 24-bit recording.

Headroom is how much space you leave between level at which you record something and the maximum level of your gear at which "clipping" occurs.

In the analog realm, you can get away with leaving very little headroom and record your signal very close to 0dB on the peaks. This is because some natural limiting occurs in analog gain stages as you exceed 0dB and unless your signal is far too hot the clipping distortion you might get on very short transients is not unpleasant sounding.

In the digital realm, however, when you exceed 0db, the tops of all wave peaks beyond 0db are simply chopped off flat, like a square wave. This results in extremely unpleasant-sounding clipping distortion. So in the digital realm, you want to leave a lot of headroom on all your initial tracks, and still a fair amount of headroom on your mixdowns, so that you run absolutely no risk of digital clipping. It's only at the final stage when you are preparing a mixdown to encode to a downloadable file format (such as MP3), or to burn to CD, that you finally increase the volume as close to 0dB as possible (usually around -0.2 or -0.3dB), with a little judicious limiting of peaks to increase the overall "loudness" slightly.

Procedure - Setting your gain stages by the numbers

Set the gain of the analog signal from the M3 into your audio interface.

Set up the combi, program, or song that you are recording (this might be played live by you, or played by recorded MIDI tracks in your external sequencer) and in the case of combis and songs, mute the specific tracks that you do not want to record.

Now play your keyboard or playback the recorded MIDI tracks while looking at the input meters on your audio interface. Find your MASTER volume control for the combi/program/song (on the M3, this is on page P9 MFX/TFX, on the Routing tab) and adust the MASTER volume so that your peaks on the input meter of your audio interface are just starting to flash red. Note: if you cannot get the input meters on your audio interface to flash red even when the MASTER volume from the M3 is maxed out at at value of 127, then you need to start increasing the input gain of your audio interface until you start seeing the red flashes.

Once you've got the peaks from the loudest sections of your music just barely flashing red on your audio interface inputs, now reduce the MASTER volume slowly until you the stop seeing red flashes on the peaks. (Or if you had to also increase the input gain on your audio interface, reduce that gain first to achieve this goal.)

Congratulations! Your analog signal now has the best possible SN ratio going into the AD converters on your audio interface.

Set the gain of the digital signal from your audio interface into the armed audio track of your external sequencer.

Be sure to set the input gain to 0dB on the track that you're going to use for recording.

If you can make the track's input meter display numbered dB values, do so.

If you can specify different types of input metering for the track's input meter, choose Peak metering.

Now play your material again and this time adust the digital output gain control of your audio interface as needed to make the track's input meter read -12dB on the highest peaks in the signal. Yes, I said -12dB. With 24-bit recording you can even go as low as -18dB with zero loss of quality, resolution, dynamic range or anything. Avoid all temptation to let those peaks go higher than -12dB. What you want on your individual tracks at this stage is lots and lots of headroom, because these signals are going to be summed together in your mixdown, and that means your Master buss is going to get much louder than -12dB. With a lot of audio tracks, you'll be turning them all down to keep your Master buss from getting too close to 0dB.

Record your audio track. Don't worry if the recorded waveform looks dinky. All is well. You don't have to worry about big, pretty waveforms until after you've finished your mixdown and have rendered a WAV or AIFF file from your external sequencer.

Repeat this entire process every time you change the source that you're recording from the M3. If you mute or unmute even one more track in the combi/song, for example, or if you change combis/programs/songs, etc., you must start from step 1.2 each and every time to get the perfect recorded analog track.

When it comes time to mixdown all your tracks, your target is now a -6dB peak reading in the Master buss on the loudest sections of your mixdown. Again, avoid the temptation to let your peaks ride higher than -6dB. Yes, your finished waveform will still look wimpy and small compared to the monster waveforms of all those modern reference CDs that you've ripped. Yes, if you encoded an MP3 or burned a CD from this -6dB mixdown, it will not sound loud enough. All is well so far.

The time to play games with the loudness of your mixdown is after you've rendered your mixdown to a 24-bit WAV or AIFF file from your external sequencer. Now you load that puppy into a good audio editing program like SoundForge or WaveLab and you normalize it to -0.2 or -0.3dB by applying some very moderate -3dB limiting to the highest peaks with the good limiters available in those programs. (Some of the "normalization" utilities will do this all for you in one easy operation.). Congratulations! Now you have a waveform that doesn't look so wimpy any more, and will be "loud enough" without buying into the downward spiral of the Loudness War. Your shit will sound better than some way overcompressed crap from a major label because it will actually have some…gasp!..dynamic range in it.

Finally, you can encode that 24-bit normalized WAV or AIFF file directly to your compressed downloadable format of choice (MP3, AAC, whatever). Or if you plan to burn a CD, you can use SoundForge or WaveLab to dither the file down to 16-bits first.

Tips

Note that if for some reason you plan to import your finished file into the M3 for use as a (multi)sample, you must dither it down to 16 bits first. And depending on the sample rate you work with, you might also need to perform sample rate conversion to 48000. Remember that the M3's internal sampler requires 16-bit, 48000 file format.

What other "experts" have to say

The debate surrounding this subject can make your head explode. You've been duly warned, lol. And it's hard to find good information about the subject. Here are some good links to other popular material on the subject.